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Is WebRTC UDP or TCP? The Real Answer

If you’ve been exploring WebRTC for real-time audio, video, and data, you may be wondering which transport protocol it uses: UDP or TCP. The truth is that WebRTC can use both — but prefers one over the other for good reasons.


🛰 Why WebRTC Favors UDP

By default, WebRTC uses UDP (User Datagram Protocol) for media transport. UDP is connectionless, which means it sends packets without waiting for acknowledgements. This makes it ideal for real-time communication where speed matters more than perfect delivery.

UDP benefits for WebRTC include:

  • Low latency — no delays from retransmission.
  • Better handling of real-time streams — packet loss is preferable to lag in a video call.
  • Optimised for live audio and video — essential for smooth playback.

If you want a deeper technical dive, the IETF’s UDP specification outlines its design.


🔄 When WebRTC Falls Back to TCP

Sometimes, UDP just isn’t possible — for example, if a firewall or network policy blocks it. In these cases, WebRTC will attempt to connect over TCP (Transmission Control Protocol) instead.

TCP is more reliable because it ensures every packet arrives, but it also:

  • Introduces extra latency.
  • Can cause “buffer bloat” during congestion.
  • Isn’t ideal for continuous real-time streams.

🌐 How WebRTC Decides Which to Use

WebRTC uses the ICE (Interactive Connectivity Establishment) framework to test different connection paths between peers. ICE works with STUN and TURN servers to:

  1. Try UDP first for lowest latency.
  2. Fall back to TCP if UDP fails.
  3. Use TURN relay servers as a last resort.

🛠 Best Practices for Choosing UDP or TCP in WebRTC

  • Prioritise UDP for calls, conferences, and live events.
  • Allow TCP as a fallback for network-restricted environments.
  • Host TURN servers in strategic locations to improve connectivity.
  • Use monitoring tools like WebRTC Internals to see which protocol is in use.

Final Thoughts

WebRTC is designed to prefer UDP for performance, but it’s versatile enough to use TCP when necessary. This adaptability is part of why it’s become the backbone of modern real-time communication.

For more articles like this one, visit SoftpageCMS for in-depth VoIP and WebRTC guides.

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